1. Field of the Invention
The invention generally relates to communications systems in which speech signals are transmitted between terminals over a network. In particular, the invention relates to systems and methods for reducing the amount of network bandwidth consumed by the transmission of speech signals between such terminals.
2. Background
The use of mobile communications has increased exponentially since the introduction of the technology just a few decades ago. The increase in users has led to the development of more and more bandwidth-efficient systems, starting with the conversion from the first generation analog-based Advanced Mobile Phone System (AMPS) cellular phone system to the second generation and beyond digital systems. As wireless carriers moved to support more and more users, the underlying speech codec has become increasingly efficient, with an approximate three-fold reduction in bandwidth. Current speech coding standards in today's mobile communications systems use 4-12 kilobits per second (kb/s) for each speech signal.
Although the advance in coding efficiency has been impressive, it is unlikely to continue and most likely is near its limit given the current set of design parameters. The performance of speech codecs can be measured by a set of attributes that include: bit rate, speech quality, degradation caused by channel impairments, delay, and computational complexity (both cycle usage and memory usage). Generally, there is a trade-off between good performance in one or more attributes and lower performance in others. The interplay between the attributes is governed by the fundamental laws of information theory, the properties of the speech signal, limitations in the equipment used, and limitations in human knowledge.
To design a speech codec, one must know the desired values for its attributes. A common approach to developing a speech codec is to constrain all attributes but one quantitatively. The design objective is then to optimize the remaining attribute (usually speech quality or bit rate) subject to these constraints.
Today's speech coding systems have been designed to minimize bit rate and maximize speech quality while maintaining limits of computational complexity, memory and storage as dictated by the economics of the terminals and the desire for smaller, sleeker handsets with longer battery life. However, as the technology used to implement these terminals continues to follow Moore's Law, the computation speed of processors continues to increase, the capacity of memory components continues to grow, and the power consumption for these devices continues to shrink. Unfortunately, however, the available bandwidth for communications remains constant.
As the use of mobile communications systems continue to grow, the pressure to increase capacity will mount. As mentioned above, given the current constraints on computational complexity and memory usage, it is unlikely that the speech signal can be compressed much further without compromising quality. However, as the capabilities of terminals and network nodes increase, the limits of various system attributes may be reconsidered and a speech codec may potentially be designed that significantly further reduces the bandwidth requirements.
For example, today's speech codecs are generally designed for speaker-independent use. However, mobile communications appears to be moving to a use-scenario in which everyone has their own cellular phone or communication device. In addition, the majority of telephone calls today are between a small set of people. These facts are not exploited at all in current speech compression schemes. The core network today is involved in call setup, call routing, billing, and the like, but is not exploited in any way to improve the efficiency of the speech codec.